Re: [rtcweb] Forking & Early Media - Was Re: Minimal SDP negotiation mechanism

"Ravindran Parthasarathi" <pravindran@sonusnet.com> Wed, 21 September 2011 06:23 UTC

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Thread-Topic: [rtcweb] Forking & Early Media - Was Re: Minimal SDP negotiation mechanism
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From: "Ravindran Parthasarathi" <pravindran@sonusnet.com>
To: "Roman Shpount" <roman@telurix.com>
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Subject: Re: [rtcweb] Forking & Early Media - Was Re: Minimal SDP negotiation mechanism
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Hi Roman,

 

Thanks for the correction. I agree with you that there is no need to
mention that forking is not support in RTCWeb. Your simple model works
for browser as RTP End-point. One problem with the generic statement of
multiple answer is that it may violate offer/answer model itself. For
example: 18x and 200 having different answer for INVITE offer (standard
implementation violation of offer/answer ;-))

 

Hope we are in the same page that forking related API and its parameters
shall be decided by W3C

 

Thanks

Partha

 

From: Roman Shpount [mailto:roman@telurix.com] 
Sent: Wednesday, September 21, 2011 7:51 AM
To: Ravindran Parthasarathi
Cc: Hadriel Kaplan; Cullen Jennings; rtcweb@ietf.org
Subject: Re: [rtcweb] Forking & Early Media - Was Re: Minimal SDP
negotiation mechanism

 

Small correction to what you are saying -- new dialogs cannot be stopped
using CANCEL (this will stop all dialogs). They can only be stopped
using BYE.

I would consider it very unfortunate if we decide not to support forking
in RTC. It needs to be handled, and not in signaling, but by defining
what to do with multiple incoming media RTP streams and multiple
answers. The simplest model would be ignore everything except the RTP
media defined in the last answer and have an ability to update the
answer to the last offer. Alternative would be to create a new
PeerConnection based on a new answer (this is API change so should
probably go to W3C).
_____________
Roman Shpount



On Tue, Sep 20, 2011 at 10:11 PM, Ravindran Parthasarathi
<pravindran@sonusnet.com> wrote:

Forking in SIP does not apply in the literal sense to lot of SIP
applications (ISDN-SIP gateway, End-point which can't perform mixing).
In case of ISDN-SIP gateway, SIP callleg has to handle all forked
dialogs till 200 OK is received from anyone of the UAS and reject all
other dialogs with CANCEL, the media plane update is depend upon the
implementation whether to override the last SDP in media plane in case
mixing is not possible. I'm saying in this mail thread to highlight
forking handling in browser (as a SIP UA application) is not an
exception and it is the decision which has to be taken by any SIP
application development (and not SIP framework) for that matter.

SIP application forking behavior depends upon RTP model (endpoint or
mixer). In case browser acts only as endpoint, I agree with Cullen that
forking shall be handled by browser without application aware and no
need of API or callback.

The counter argument may be that my innovative mixing application in
browser is stopped by this API model.  In the generic SIP framework, the
callbacks are provided to handle this situation, default callback
function (browser as endpoint) are provided to reduce the application
awareness. From the API perspective, offer & answer state machine is not
required to be handled in application but we required to know whether
the application prefers which media model whether as end-point or mixer
and let end-point model be default. IMO, it is browser API design which
belongs to W3C. Please correct me here in case this API design is
somehow related to IETF.

Thanks
Partha

>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
Behalf
>Of Hadriel Kaplan
>Sent: Wednesday, September 21, 2011 4:06 AM
>To: Cullen Jennings
>Cc: rtcweb@ietf.org
>Subject: Re: [rtcweb] Forking & Early Media - Was Re: Minimal SDP
>negotiation mechanism
>
>
>On Sep 20, 2011, at 4:07 PM, Cullen Jennings wrote:
>
>> That said, I think that doing both forking and early media is hard.
>Lets assume we are using a signaling gateway that is not a media
gateway
>to translate between a SIP call on one side and whatever is happening
>over on the browser side. The basic issue is the browser initiating the
>communications needs to be able to start receiving multiple RTP streams
>before it even has signaling information to tell it how many it might
>receive.
>
>Not really - there will be signaling, because there has to be SDP
>answers even just to get ICE to work before the media starts flowing in
>many NAT cases.  And even in practice in SIP there're usually SDP
>answers in 18x to open "gates", and to get upstream DTMF.  So if the
>concern is just that there's no signaling to tell the browser there are
>multiple RTP streams coming, I think that can be allayed.
>
>The really hard part is knowing which stream to use/render/send-to,
>imho.  And putting that decision in the gateway isn't good - the best
>decider of that is probably the JS in the browser.
>
>
>> To simplify this problem, Cary and my draft proposes not allowing
>forking on the SIP side of the signaling gateway but still allowing
>early media. If you wanted to do do forking in this case, one would
need
>a SBC that processed media and turned the forked medial legs into one
>media leg.
>
>Obviously you can request that a request not be forked, using caller-
>prefs, but you can't "not allow" forking on the SIP side.  That would
>make it not SIP.  I know forking is hard, but that's life.  It's not
>appropriate for this WG to make fundamental changes/limitations to the
>SIP protocol, just because some of it's "hard" for a browser.
>
>-hadriel
>
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